Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls.

webrtc/media/base/fakemediaengine.h

Index: webrtc/media/base/fakemediaengine.h
diff --git a/webrtc/media/base/fakemediaengine.h b/webrtc/media/base/fakemediaengine.h
index c0aacfce5c19e6535c169096d859d3837062b3ee..4984d1b88c3168dbcaff6ef43501ab5b07292b58 100644
--- a/webrtc/media/base/fakemediaengine.h
+++ b/webrtc/media/base/fakemediaengine.h
@@ -604,6 +604,7 @@ class FakeVideoMediaChannel : public RtpHelper<VideoMediaChannel> {
     return true;
   }
 
+  void FillBitrateInfo(BandwidthEstimationInfo* bwe_info) override {}
   bool GetStats(VideoMediaInfo* info) override { return false; }
 
  private: